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Cisco IOS SIP Configuration GuideDialpeer Configuration

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Terminology

• Call - A connection terminating on or passing through agateway.

• Call Leg - The segment of a call associated with a particularsignaling and transport technology, for example SIP or PSTN

• Service Provider - the implementation of the Interface for aparticular protocol (signaling stack)

• Interface (voice-port) - A physical or logical connector thatcarries call legs. For example, an analog line or a T1/PRI span.The IP network is also modeled as an interface.

• Application (a.k.a. Session application) - accepts and createscall-legs, provides feature platform.

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Dial Peer

• A dial-peer is the entity to which a call isconnected. Includes VoIP, Pots etc.

• Incoming dial-peers point to anapplication to handle an incoming call

• Outgoing dial-peers pick an interface,PSTN or SIP, to handle an outgoing call.

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VoIP DialPeer

• Map phone numbers (E.164 addresses) or SIP URIs to IPaddresses or DNS names

• Describe transport characteristics of the connection like: codec,vad, QoS, dtmf-relay type etc.

• Example: dial-peer voice 111 voip destination-pattern 60154 incoming called number 1001 session protocol sipv2 session target dns:sipserver1.hawaii.edu dtmf-relay rtp-nte codec g711ulaw

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URI Matching

• From 12.3(4)T onwards, a voip dialpeercan be matched based on a sip: uri

• A voice class uri needs to be configured:

voice class uri SIP_1 sip

user abc

host sip.com

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URI Matching contd…

• On the dialpeer, the voice class needs tobe associated with from, to or requesturi.

dial-peer voice 111 voip destination-pattern 60154 incoming called number 1001 incoming uri from SIP_1 session protocol sipv2 session target dns:sipserver1.hawaii.edu ….

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VoIP Dialpeer Matching Rule

• Inbound dialpeer incoming uri request incoming uri to incoming uri from incoming called-number answer address destination-pattern

• Outbound dialpeer destination-uri destination-pattern

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POTS Dialpeer

• Map phone numbers to voice ports.• Destination-pattern is used to match an outbound dialpeer,

incoming called-number is used to match an inbounddialpeer

• Example: dial-peer voice 100 pots destination-pattern 9000 port 1/0/0• Voice ports further specify signaling properties

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Order of Dialpeer matching

• All matched dialpeer are sorted based onpreference. Higher preference is given todialpeers with an exact pattern match.

• Two dialpeers with the same pattern match willbe tried in the order they were configured.

• preference command can be used to break thetie between two dialpeers with same matchcharacteristics.

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Number Translation using TranslationProfile

• Voice Translation Profiles introduce a scheme to translatenumbers.

• The translation rules replace a sub string of the input number ifthe number matches the match pattern, number plan, and typepresent in the rule.

• Called, Calling and Redirect-Called numbers can be defined in atranslation profile. Each type of call number in the profile canhave different translation rules.

• Translation profiles can be referenced on: Trunk Group, SourceIP Group, Dial-Peer, Voice-Port, VoIP Incoming

• The voice translation rules use characters similar to RegularExpression Syntax (regexp)

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Configuring Translation Rule

• Syntax: Router(config)# voice translation-rule <num> Router(cfg-translation-rule)# rule precedence /match-pattern/ /replace-pattern/ [type {match-type

replace-type} [plan {match-type replace-type}]]• Examples: 1. This example replaces any occurrence of the number "123" with "456". voice translation-rule 1 rule 1 /123/ /456/ 2. Match 1# at the beginning and replace it with Null. voice translation-rule 2 rule 2 /^1#/ // 3. Expand 5 digit number to 10 digits voice translation-rule 3 rule 3 /25555/ /91939&/

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Configuring Translation Profile

• Once a translation rule has been configured, translation profile can be configured by: voice translation-profile <name> translate called <translation-rule num> translate calling <translation-rule num> translate redirect-called <translation-rule num>• Dial-Peer configuration: dial-peer voice <num> [pots|voip] translation-profile [incoming | outgoing] <name>• For more information on number translation:

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00803f818a.shtml

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Cisco IOS SIP Configuration GuideSIP Feature Configuration

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Reliable Provisional Response

• Gateway can be configured to send 18x response reliably as inRFC 3262.

• Global configuration is under voice-service voip; sip. It canalso be configured on the voip dialpeer. Dialpeer configurationwill take precedence over global configuration

• To configure it: router# voice-service voip router(conf-voi-serv)#sip

router(conf-serv-sip)# rel1xx [require|supported] 100rel• Default mode is rel1xx supported 100rel

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Codec configuration

• Codec can be configured on the voip dialpeer usingcodec <codec> cli.

Example: router# conf t router(config)#dial-peer voice 6 voip router(config-dial-peer)#codec g711ulaw• Codecs configured on the outbound dialpeer will be

sent in sdp of INVITE. Default codec is G729

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Codec Configuration contd ..

• More than one codec can be configured using voice-classcodec.

Example: router# conf t router(config)#voice class codec <num> router(config-class)#codec preference 1 g711alaw router(config-class)#codec preference 2 g711ulaw On the dialpeer: router(config)#dial-peer voice 6 voip router(config)# voice-class codec <num>

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Configuration under sip-ua

• Configurations specific to sip user agent are under sip-ua.Commonly used configs are message retry count, retryinterval configs, configuring an outbound server

• Configuring number of retries. router(config)# sip-ua router(config-sip-ua)# retry <message> <number>• Signaling timer configuration. router(config)# sip-ua router(config-sip-ua)# timers <message> <timer-val>

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sip-ua configurations contd ..

• Configuring an outbound server

router(config)# sip-ua

router(config-sip-ua)# sip-server <server address>

On the outbound voip dialpeer:

router(config)#dial-peer voice 6 voip

router(config)# session-target sip-server

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sip-ua Configuration contd …

• Overriding default SIP-PSTN disconnect cause code router(config)# sip-ua router(config)# set pstn-cause <num> sip-status

<num> router(config)# set sip-status <num> pstn-status

<num> Range of sip-status is 400-699 Range of pstn-status is 1-127

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Caller identity and Privacy

• IOS SIP gateway uses Remote-Party-ID header that identifiesthe calling party and carries presentation and screeninginformation.

• Implementation is based on draft-ietf-privacy-.02.txt, SIPExtensions for Caller Identity and Privacy.

• For PSTN-SIP call, information from octet3a is used tocreate presentation and screening parameters in Remote-Party-ID header.

• For SIP-PSTN, presentation and screening parameters inRemote-Party-ID header is used to create octet3ainformation in ISDN SETUP.

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Caller Identity and Privacy contd..

• Additional CLI commands allow alternative callinginformation treatments for calls entering the SIP trunkinggateway. Configurable treatment options for SIP-PSTN:

• Calling name and number pass-through (default). • No calling name or number sent in the forwarded Setup

message. • Calling name unconditionally set to the configured string

in the forwarded Setup message. • Calling number unconditionally set to the configured string

in the forwarded Setup message.

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Caller Identity and Privacy contd…

• Configurable treatment options for PSTN-SIP: • Calling name and number pass-through (default). • No calling name or number sent in the forwarded INVITE message. • Display-name of the From header unconditionally set to the configured

string in the forwarded INVITE message. • User part of the From header unconditionally set to the configured string

in the forwarded INVITE message. • Display-name of the Remote-Party-ID header unconditionally set to the

configured string in the forwarded INVITE message. • User part of the Remote-Party-ID header unconditionally set to the

configured string in the forwarded INVITE message.• P-Asserted-Identity support will be available in a future release.

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Addition SIP gateway features

• Call Transfer• T.38 fax with fallback to fax-passthrough• Buffered Calling-Name• Registration• Digest Authentication• Call Redirection• Ability to configure source address for signaling and media